[Libav-user] Problem with compressing AAC files with sampling rate 16000

adev dev androiddevmar11 at gmail.com
Thu Sep 18 14:43:18 CEST 2014

I am compressing movies from bitmaps and AAC files. Normally AAC files are
recorded with following params: bit rate: 192000, sampling rate 44100.
These audio params are set in output context (AVFormatContext) in
compression. In happy day scenario it is working correctly. Movie and sound
are correct.

The problem occures if sound params are "strange". From time to time I have
AAC files recorded with params: bitrate 96000, sampling rate 16000. In this
case sound is very fast and ends before end of movie. If I set these params
(96000, 16000) in output context there is no sound in the movie at all.
Created movie file has audio stream with bitrate about 500 b/s what is
really strange.

What else has to be set in FFMPEG to correctly compress AAC files with bit
rate 96000 and sampling rate 16000?

I am using FFMPEG 2.1 and native AAC encoder.
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